// VideoScaleTest.cpp : Defines the entry point for the console application.
// src code is come from ffmpeg realease "./share/ffmpeg/examples/resampling_audio.c"
// compile in vs2013 with multi-byte character set
// Modify by Tocy <zyvj@qq.com>.

#include "stdafx.h"

extern "C"
{
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
}

#pragma comment(lib, "libavutil.a")
#pragma comment(lib, "libswresample.a")


static int get_format_from_sample_fmt(const char **fmt, enum AVSampleFormat sample_fmt)
{
	int i;
	struct sample_fmt_entry {
		enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
	} sample_fmt_entries[] = {
		{ AV_SAMPLE_FMT_U8, "u8", "u8" },
		{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
		{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
		{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
		{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
	};
	*fmt = NULL;

	for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
		struct sample_fmt_entry *entry = &sample_fmt_entries[i];
		if (sample_fmt == entry->sample_fmt) {
			*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
			return 0;
		}
	}

	fprintf(stderr,
		"Sample format %s not supported as output format\n",
		av_get_sample_fmt_name(sample_fmt));
	return AVERROR(EINVAL);
}

/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
	int i, j;
	double tincr = 1.0 / sample_rate, *dstp = dst;
	const double c = 2 * M_PI * 440.0;

	/* generate sin tone with 440Hz frequency and duplicated channels */
	for (i = 0; i < nb_samples; i++) {
		*dstp = sin(c * *t);
		for (j = 1; j < nb_channels; j++)
			dstp[j] = dstp[0];
		dstp += nb_channels;
		*t += tincr;
	}
}

// usage: <exe_path> <output_file> <outsize>
// as:	  "AudioResampleTest.exe aout"
int _tmain(int argc, _TCHAR* argv[])
{
	int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
	int src_rate = 48000, dst_rate = 44100;
	uint8_t **src_data = NULL, **dst_data = NULL;
	int src_nb_channels = 0, dst_nb_channels = 0;
	int src_linesize, dst_linesize;
	int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
	enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
	const char *dst_filename = NULL;
	FILE *dst_file;
	int dst_bufsize;
	const char *fmt;
	struct SwrContext *swr_ctx;
	double t;
	int ret;

	if (argc != 2) {
		fprintf(stderr, "Usage: %s output_file\n"
			"API example program to show how to resample an audio stream with libswresample.\n"
			"This program generates a series of audio frames, resamples them to a specified "
			"output format and rate and saves them to an output file named output_file.\n",
			argv[0]);
		exit(1);
	}
	dst_filename = argv[1];

	dst_file = fopen(dst_filename, "wb");
	if (!dst_file) {
		fprintf(stderr, "Could not open destination file %s\n", dst_filename);
		exit(1);
	}

	/* create resampler context */
	swr_ctx = swr_alloc();
	if (!swr_ctx) {
		fprintf(stderr, "Could not allocate resampler context\n");
		ret = AVERROR(ENOMEM);
		goto end;
	}

	/* set options */
	av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
	av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
	av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);

	av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
	av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
	av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);

	/* initialize the resampling context */
	if ((ret = swr_init(swr_ctx)) < 0) {
		fprintf(stderr, "Failed to initialize the resampling context\n");
		goto end;
	}

	/* allocate source and destination samples buffers */

	src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
	ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
		src_nb_samples, src_sample_fmt, 0);
	if (ret < 0) {
		fprintf(stderr, "Could not allocate source samples\n");
		goto end;
	}

	/* compute the number of converted samples: buffering is avoided
	* ensuring that the output buffer will contain at least all the
	* converted input samples */
	max_dst_nb_samples = dst_nb_samples =
		av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

	/* buffer is going to be directly written to a rawaudio file, no alignment */
	dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
	ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
		dst_nb_samples, dst_sample_fmt, 0);
	if (ret < 0) {
		fprintf(stderr, "Could not allocate destination samples\n");
		goto end;
	}

	t = 0;
	do {
		/* generate synthetic audio */
		fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);

		/* compute destination number of samples */
		dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
			src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
		if (dst_nb_samples > max_dst_nb_samples) {
			av_freep(&dst_data[0]);
			ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
				dst_nb_samples, dst_sample_fmt, 1);
			if (ret < 0)
				break;
			max_dst_nb_samples = dst_nb_samples;
		}

		/* convert to destination format */
		ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
		if (ret < 0) {
			fprintf(stderr, "Error while converting\n");
			goto end;
		}
		dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
			ret, dst_sample_fmt, 1);
		if (dst_bufsize < 0) {
			fprintf(stderr, "Could not get sample buffer size\n");
			goto end;
		}
		printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
		fwrite(dst_data[0], 1, dst_bufsize, dst_file);
	} while (t < 10);

	if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
		goto end;
	fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
		"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
		fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);

end:
	fclose(dst_file);

	if (src_data)
		av_freep(&src_data[0]);
	av_freep(&src_data);

	if (dst_data)
		av_freep(&dst_data[0]);
	av_freep(&dst_data);

	swr_free(&swr_ctx);
	return ret < 0;
}

